1. Field of the Invention
The present invention relates to a microphone array system, in particular, a microphone array system including three-dimensionally arranged microphones that estimates a sound to be received in an arbitrary position in a space by received sound signal processing and can estimate sounds in a large number of positions with a small number of microphones.
2. Description of the Related Art
Hereinafter, a sound estimation processing technique using a conventional microphone array system will be described.
A microphone array system includes a plurality of microphones arranged and performs signal processing by utilizing a sound signal received by each microphone. The object, configuration, use and effects of the microphone array system vary depending on how the microphones are arranged in a sound field, what kind of sounds the microphones receive, or what kind of signal processing is performed. In the case where a plurality of sound sources of a desired signal and noise are present in a sound field, high quality enhancement of the desired sound and noise suppression are important issues to be addressed for the processing of the sounds received by microphones. In addition, the detection of the position of the sound source is useful to various applications such as teleconference systems, guest-reception systems or the like. In order to realize processing for enhancing a desired signal, suppressing noise and detecting sound source positions, it is effective to use the microphone array system.
In the prior art, for the purpose of improving the quality of the enhancement of a desired signal, the suppression of noise, and the detection of a sound source position, signal processing has been performed with an increased number of microphones constituting the array so that more data of received sound signals can be acquired. FIG. 14 shows a conventional microphone array system used for desired signal enhancement processing by synchronous addition. The microphone array system shown in FIG. 14 includes real microphones MIC0 to MICnxe2x88x921, which are arranged in an array shown as 141, delay units D0 to Dnxe2x88x921 for adjusting the timing of signals of sounds received by the respective real microphones 141, and an adder 143 for adding signals of sounds received by the real microphones 141. In the enhancement of a desired sound according to this conventional technique, a sound from a specific direction is enhanced by adding plural received sound signals that are elements for addition processing. In other words, the number of sound signals used for synchronous addition signal processing is increased by increasing the number of the real microphones 141 so that the intensity of a desired signal is raised. In this manner, the desired signal is enhanced so that a distinct sound is picked out. As for noise suppression, synchronous subtraction is performed to suppress noise. As for the detection of the position of a sound source, synchronous addition or the calculation of cross-correlation coefficients is performed with respect to an assumed direction. In these cases as well, the quality of the sound signal processing is improved by increasing the number of microphones.
However, this technique for microphone array signal processing by increasing the number of microphones is disadvantageous in that a large number of microphones should be prepared to realize high quality sound signal processing, so that the microphone array system results in a large scale. Moreover, in some cases, it may be difficult to arrange microphones in number necessary for sound signal receiving of required quality in a necessary position physically because of spatial limitation.
In order to solve the above problems, it is desired to estimate a sound signal that would be received in an assumed position based on actual sound signals received by actually arranged microphones, rather than receiving a sound by microphones that are arranged actually. Furthermore, using the estimated signals, the enhancement of a desired signal, noise suppression and the detection of a sound source position can be performed.
The microphone array system is useful in that it can estimate a sound signal to be received in an arbitrary position on an array arrangement, using a small number of microphones. The microphone array system is preferable, in that it can estimate a sound signal to be received in an arbitrary position in a three-dimensional space, because sounds are propagated actually in the three-dimensional space. In other words, it is required not only to estimate a sound signal to be received in an assumed position on the extended line (one-dimensional) of a straight line on which a small number of microphones are aligned, but also to estimate with respect to a signal from a sound source that is not on the extended line while reducing estimation errors. Such high quality sound signal estimation is desired.
Furthermore, it is desired to develop an improved signal processing technique for signal processing procedures that are applied to the sound signal estimation so as to improve the quality of the enhancement of a desired sound, the noise suppression, the sound source position detection.
Therefore, with the foregoing in mind, it is an object of the present invention to provide a microphone array system with a small number of microphones arranged three-dimensionally that can estimate a sound signal to be received in an arbitrary position in the three-dimensional space with the small number of microphones.
Furthermore, it is another object of the present invention to provide a microphone array system that can perform sound signal estimation of high quality, for example by performing interpolation processing for predicting and interpolating a sound signal to be received in a position between a plurality of discretely arranged microphones, even if the number of microphones or the arrangement location cannot be ideal.
Furthermore, it is another object of the present invention to provide a microphone array system that realizes estimation processing that is better in sound signal estimation in an arbitrary position in the three-dimensional space than sound signal estimation processing used in the conventional microphone array system, and can perform sound signal estimation of high quality.
A microphone array system of the present invention includes a plurality of microphones and a sound signal processing part. As for the microphones, at least three microphones are arranged on each spatial axis. The sound signal processing part estimates a sound signal in an arbitrary position in a space by estimating a sound signal to be received at each axis component in the arbitrary position, utilizing the relationship between the difference, which is a gradient, between neighborhood points on the time axis of the sound pressure of a received sound signal of each microphone and the difference, which is a gradient, between neighborhood points on the spatial axis of the air particle velocity, and the relationship between the difference, which is a gradient, between neighborhood points on the spatial axis of the sound pressure and the difference, which is a gradient, between neighborhood points on the time axis of the air particle velocity, and based on the temporal variation of the sound pressure and the spatial variation of the air particle velocity of the received sound signal of each microphone arranged in each spatial axis direction; and synthesizing the estimated signals three-dimensionally.
This embodiment makes it possible to estimate a sound signal in an arbitrary position in a space by utilizing the relationship between the gradient on the time axis of the sound pressure calculated from the temporal variation of the sound pressure of a sound signal received by each microphone and the gradient on the spatial axis of the air particle velocity calculated based on a received signal between the microphones arranged on each axis.
Furthermore, a microphone array system of the present invention includes a plurality of microphones and a sound signal processing part. The microphones are arranged in such a manner that at least three microphones are arranged in a first direction to form a microphone row, at least three rows of the microphones are arranged so that the microphone rows are not crossed each other so as to form a plane, and at least three layers of the planes are arranged three-dimensionally so that the planes are not crossed each other, so that the boundary conditions for the sound estimation at each plane of the planes constituting the three dimension can be obtained. The sound signal processing part estimates a sound in each direction of a three-dimensional space by estimating sound signals in at least three positions along a direction that crosses the first direction, utilizing the relationship between the difference, which is a gradient, between neighborhood points on the time axis of the sound pressure of a received sound signal of each microphone and the difference, which is a gradient, between neighborhood points on the spatial axis of the air particle velocity, and the relationship between the difference, which is a gradient, between neighborhood points on the spatial axis of the sound pressure and a difference, which is a gradient, between neighborhood points on a time axis of the air particle velocity, and based on the temporal variation of the sound pressure and the spatial variation of the air particle velocity of received sound signals in at least three positions aligned along the first direction; and further estimating a sound signal in the direction that crosses the first direction based on the estimated signals in the three positions.
This embodiment provides the boundary conditions for the sound estimation at each plane of the planes constituting the three dimension, so that a sound signal in an arbitrary position in the three-dimensional space can be estimated by utilizing the relationship between the gradient on the time axis of the sound pressure calculated from the temporal variation of the sound pressure of a sound signal received by each microphone and the gradient on the spatial axis of the air particle velocity calculated based on a received signal between the microphones arranged on each axis.
Furthermore, a microphone array system of the present invention includes a plurality of directional microphones and a sound signal processing part. As for the directional microphones, at least two directional microphones are arranged with directivity on each spatial axis. The sound signal processing part estimates a sound signal in an arbitrary position in a space by estimating a sound signal to be received at each axis component in the arbitrary position utilizing the relationship between the difference, which is a gradient, between neighborhood points on the time axis of the sound pressure of a received sound signal of each microphone and the difference, which is a gradient, between neighborhood points on the spatial axis of the air particle velocity, and the relationship between the difference, which is a gradient, between neighborhood points on the spatial axis of the sound pressure and the difference, which is a gradient, between neighborhood points on the time axis of the air particle velocity, and based on the temporal variation of the sound pressure and the spatial variation of the air particle velocity of a received sound signal of each of the directional microphones arranged in each spatial axis direction; and synthesizing the estimated signals three-dimensionally.
This embodiment makes it possible to estimate a sound signal in an arbitrary position in a space by utilizing the gradient on the time axis of the sound pressure calculated from the temporal variation of the sound pressure of a sound signal received by each directional microphone, the gradient on the spatial axis of the air particle velocity calculated based on a received signal between the directional microphones arranged so that the directivities thereof are directed to the respective axes, and the correlation thereof.
Next, a microphone array system of the present invention includes a plurality of directional microphones and a sound signal processing part. The directional microphones are arranged in such a manner that at least two directional microphones are arranged with directivity to a first direction to form a microphone row, at least two rows of the directional microphones are arranged so that the microphone rows are not crossed each other so as to form a plane, and at least two layers of the planes are arranged three-dimensionally so that the planes are not crossed each other, so that the boundary conditions for the sound estimation at each plane of the planes constituting the three dimension can be obtained. The sound signal processing part estimates a sound in each direction of the three-dimensional space by estimating sound signals in at least two positions along a direction that crosses the first direction, utilizing the relationship between a difference, which is a gradient, between neighborhood points on the time axis of the sound pressure of a received sound signal of each microphone and the difference, which is a gradient, between neighborhood points on the spatial axis of the air particle velocity, and the relationship between the difference, which is a gradient, between neighborhood points on the spatial axis of the sound pressure and the difference, which is a gradient, between neighborhood points on the time axis of the air particle velocity, and based on the temporal variation of the sound pressure and the spatial variation of the air particle velocity of received sound signals in at least two positions aligned along the first direction; and further estimating a sound signal in the direction that crosses the first direction based on the estimated signals in the two positions.
This embodiment provides the boundary conditions for the sound estimation at each plane of the planes constituting the three dimension, and makes it possible to estimate a sound signal in an arbitrary position in the three-dimensional space by utilizing the gradient on the time axis of the sound pressure calculated from the temporal variation of the sound pressure of a sound signal received by each directional microphone, the gradient on the spatial axis of the air particle velocity calculated based on a received signal between the directional microphones arranged so that the directivities thereof are directed to respective axes, and the correlation thereof.
In the microphone array system, it is preferable that the relationship between the gradient on the time axis of the sound pressure and the gradient on the spatial axis of the air particle velocity of the received sound signal is expressed by Equation 2:
(vx(xi+1,yj,zg,tk)xe2x88x92
vx(xi,yj,zg,tk))+
(vy(xi,yj+1,zg,tk)xe2x88x92
vy(xi,yj,zg,tk))+
(vz(xi,yj,zg+1,tk)xe2x88x92
vz(xi,yj,zg,tk)=
b(p(xi+1,yj+1,zg+1,tk+1)xe2x88x92
p(xi+1,yj+1,zg+1,tk))xe2x80x83xe2x80x83Equation 2
where x, y, and z are spatial axis components, t is a time component, v is the air particle velocity, p is the sound pressure, and b is a coefficient.
In the microphone array system, it is preferable that the sound signal processing part includes a parameter input part for receiving an input of a parameter that adjusts the signal processing content. One example of an input parameter is a sound signal enhancement direction parameter for designating a specific direction in which sound signal estimation is enhanced is supplied to the parameter input part, thereby enhancing a sound signal from a sound source in the specific direction. Another example of an input parameter is a sound signal attenuation direction parameter for designating a specific direction in which sound signal estimation is reduced is supplied to the parameter input part, thereby removing a sound signal from a sound source in the specific direction.
This embodiment makes it possible for a user to adjust and designate the signal processing content in the microphone array system.
In the microphone array system, it is preferable that the interval distance between adjacent microphones of the arranged microphones is within an interval distance that satisfies the sampling theorem on the spatial axis for the frequency of a sound signal to be received.
This embodiment makes it possible to perform high quality signal processing in a necessary frequency range by satisfying the sampling theorem.
In the microphone array system, it is preferable that the sound signal processing part includes a band processing part for performing band division processing and frequency shift for band synthesis for a received sound signal at the microphones.
This embodiment makes it possible to adjust the apparent bandwidth of a signal and shift the frequency of the signal received by the microphones, so that the same effect as that obtained by adjusting the sampling frequency of the signal received by the microphones can be obtained.
Furthermore, a microphone array system of the present invention includes a plurality of microphones and a sound signal processing part. As for the microphones, a plurality of microphones are arranged in three orthogonal axis directions in a predetermined space. The sound signal processing part connected to the microphones estimates a sound signal in an arbitrary position in a space other than the space where the microphones are arranged based on the relationship between the positions where the microphones are arranged and the received sound signals.
This embodiment makes it possible to estimate a sound signal in an arbitrary position in a space other than the space where the microphones are arranged.
In the microphone array system, it is preferable that the microphones are mutually coupled and supported on a predetermined spatial axis.
Preferably, this support member has a thickness of less than xc2xd, preferably less than xc2xc, of the wavelength of the maximum frequency of the received sound signal, and preferably this support member is solid, and is hardly oscillated by the influence of the sound.
This embodiment makes it possible to provide a microphone array system where the microphones are arranged actually in a predetermined position interval distance, and the oscillation by the sound can be suppressed so as to reduce noise to the received signal.
These and other advantages of the present invention will become apparent to those skilled in the art upon reading and understanding the following detailed description with reference to the accompanying figures.